TrixBox SIP Peer and it's extension

Question

I am totally new to this PBX, I really don't know what is it and I wanna learn it ASAP, I've created an SIP peer and an SIP user, in Extensions.conf I need the code to route all outbound calls to that SIP Peer (which is a gsm gateway of GoIP). Here're my files
Extensions.conf :

[goip]
exten => 1,1,Wait(1)
exten => 1,n,Answer
exten => 1,n,Playback(hello-world)
exten => 1,n,Wait(2)
exten => 1,n,Hangup

User Configuration in Sip_additional.conf:

[400]
deny=0.0.0.0/0.0.0.0
type=friend
secret=aneestech01
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=400@default
host=dynamic
dtmfmode=rfc2833
dial=SIP/400
context=from-internal
canreinvite=no
callgroup=
callerid=device <400>
accountcode=
call-limit=50

Peer Configuration in SIP.conf:

[gsm1]
canreinvite=no
context=goip
dtmfmode=rfc2833
fromuser=gsm1
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=g729
insecure=port,invite
secret=gsm1
type=friend
defaultname=gsm1

Please note that both my GSM Gateway and My SIP client are connected to the asterisk pbx successfully, The thing I request now is that whenever I dial a number from sip client, it should be dialed from the gsm gateway, if you need any kind of additional info please post a comment and you'll get a reply soon

thanks but the problem has already been solved below I have posted the answer which worked for me, I wanna ask another question but StackOverFlow wont let me :\ Can you help me with that please? — user2466636

arheops arheops · Accepted Answer · 2013-07-10T01:00:00

You need context like:

[from-internal]

exten => _X.,1,Dial(SIP/gsm1,,)

But i highly recomend you read some book for beginners like "asterisk the future of telephony"

TrixBox SIP Peer and it's extension

2 Answers